In the prior art, various methods and systems have been developed to decode composite signals with varying degrees of success.
One such method of decoding involves first passing an analog composite stereo signal through a low pass filter to remove the subsidiary signals, such as the SCA signal, leaving only the basic stereo signal. The basic stereo signal is then mixed with the 38 KHz subcarrier with one resulting component being one sideband of the (L-R) signal translated down to baseband. The pre-mixing basic stereo signal and the stereo signal mixed with the stereo subcarrier are in parallel passed through a low pass filter and then to a summing circuit and a subtracting circuit, the summing circuit adding the two signals together and the subtracting circuit subtracting the mixed signal from the unaltered basic stereo signal. One of the resulting components from the summation is 2*l(t) (i.e. twice the time varying value of the left(t) channel component). As a result of the subtraction, one of the resulting components is 2*r(t) (i.e. twice the time varying value of the right channel component). The right and left channel information is then easily extracted by filtering out the remaining components resulting from the mixing, summation and subtraction operations. This analog approach is well known in the art and is susceptible to all of the disadvantages inherent with analog signal processing such, problems with noise, drift with temperature, and overall circuit complexity.
Another approach was digital decoding to overcome the disadvantages inherent with analog decoding circuitry. This method involves converting the composite FM stereo signal output from the FM discriminator to digital from analog. In this instance, the 38 KHz modulated portions of composite signal are sampled at selected points when the term [sin 2.OMEGA..sub.p t] (or alternatively sin .OMEGA..sub.sc t, where .OMEGA..sub.sc is the angular frequency of the subcarrier, typically 38 KHz) in fm(t)=[l(t)+r(t)]+A.sub.p sin(.OMEGA..sub.p t)+[l(t)-r(t)]sin(2.OMEGA..sub.p t) where:
is equal to plus or minus one (the ninety degree points on the stereo subcarrier) such that the composite signal is equal to either twice the left channel (2l) or twice the right channel (2r), the left and right channel information can then be easily extracted. The substantial difficulty with this approach is that, if the samples vary from the ninety degree points on the subcarrier, the sine of the subcarrier signal will not equal plus or minus one and a given sample will not represent a signal which is essentially purely right channel information or purely left channel information; the result is a deterioration in channel separation. One means of overcoming this problem is to use a voltage controlled oscillator feedback path to phase lock the sampling frequency to the pilot signal frequency. The 19 KHz pilot signal is then used to determine when sampling of the 38 KHz modulated information will occur. This method however requires substantially complex and costly hardware to implement.
Thus, the need has arisen for improved devices, systems and methods for decoding composite signals. Such devices, systems and methods would overcome the substantial technical disadvantages inherent with currently available analog decoding means and the substantial cost and complexity disadvantages inherent with currently available digital decoding means.
Furthermore, a need has arisen to obtain a solution which avoids expensive phase locked loop circuitry and associated oscillators for decoding the left and right stereo channel signals.